[Asterisk-Users] PRI error (HDLC Bad FCS)
Hi, We currently have an Asterisk box (P4 2.4 Ghz, 512Mb RAM, 1 T100P, Zhone Channel Bank and 1 E100P) connected to an ISDN running without any problems. That machine is working for about 1 year. Two days ago, we decided to switch that machine for two PowerEdge 600SC (HA) and we got some problems. The running machine (P4 2.4Ghz) has an Intel motherboard with four 32-bit 3.3v PCI slots, which the T100P and the E100P are connected to. Great! No problem for almost a year. Our PE 600 also has an Intel motherboard which only has one PCI 32-bit 3.3v slot and four 64-bit 3.3v PCI slots. First, we tried to get one of the PE 600 working, putting the Digium cards on it. We've put the T100P on the 32-bit slot (which is the slot 0) and the the E100P on the first 64-bit slot. Searching for the T100P and E100P documentation, I've found only that the T410P works fine on 64-bit 3.3v PCI slots, but could not find any saying that wouldn't on the 64-bit 3.3v PCI slots. After loading the module (wct1xxp) and starting the Asterisk, everything seems to be right. But after 3 or 4 calls, the problems begins. During the conversation, you can hear pop's on the call, and most cases, we lost the call. Looking at the messages file, the Asterisk reports: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2. Every time Asterisk reports this error message, that problem happens again. We've tried switching the card's slot, but did not work. Switching back to the old machine, everything works fine. Some Info: PowerEdge 600SC with kernel 2.4.27 (non-SMP). Zaptel.conf # Channel Bank span=1,0,0,esf,b8zs # E1 span=2,1,0,ccs,hdb3,crc4 Zapata.conf switchtype=euroisdn signalling = pri_cpe Do you guys need more info? Any other log messages? thank you --oz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to block incoming collect calls
Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to block incoming collect calls
All right Steve. I'll ask them.. But if anybody knows that, please post an answer to the list. This is a very important Asterisk security configuration to avoid people call you without having to pay the call.. thank you Oz From: Steve Totaro [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Need to block incoming collect calls Date: Sat, 24 Jul 2004 11:57:05 -0400 I dont know about blocking in * but you should be able give the telco a call and tell them no collect calls. - Original Message - From: Osvaldo Mundim Junior [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 10:06 AM Subject: [Asterisk-Users] Need to block incoming collect calls Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] collect calls
Is it possible to set in Asterisk? Not to accept collect calls? Oz From: Osvaldo Mundim Junior [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] collect calls Date: Mon, 19 Jul 2004 16:33:19 -0300 Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] collect calls
Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do_monitor: Bad file descriptor
Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do_monitor warning message
Hi, I'm using Asterisk version (Asterisk CVS-05/30/04-16:28:04) on Debian Woody. Sometimes I get this warning message: Jun 14 13:32:41 WARNING[10251]: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When that is happening, Asterisk gets slow and close all remote active connections (asterisk -vvvcr). VoIP call alse gets bad at this time. That message appears many times, like if Asterisk were in loop. I've downloaded the version from the CVS (cvs checkout -r 1-0_stable). And I'm not using any monitor application.. Connected to this server, I have a Zhone (16 FXS extensions and 8 FXO lines) and a T100P. I'm also using the last CVS version for Zaptel and Libpri. Did anybody get the same message? How can I fix that? regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI + g729A
Hello I have the follow situatuion: ISDN | | V E100P || IAX2 / g729A || T100P | Asterisk1 |- - - - - - - - - - - - - - | Asterisk2 | - - - - - - |--| | | | | | Zhone| - - --- Here's the situation: I receive calls from the PSTN in Asterisk1 and forward the call to Asterisk2 (which is connected to a Zhone 100 channel bank). In Asterisk2 I have an AGI application written in C which does some checks before dial one of the extensions. From the AGI program, after done all I have to do, I need to call one extension, but when I call the Dial application, my AGI program exits and Asterisk gives me: -- Executing Goto([EMAIL PROTECTED]/16386, 100|1) in new stack -- Goto (default,100,1) -- Executing AGI([EMAIL PROTECTED]/16386, exm) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exm -- AGI Script Executing Application: (DIAL) Options: (Zap/23|12|m) Jun 7 13:28:30 NOTICE[730128]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- AGI Script exm completed, returning 0 -- Hungup '[EMAIL PROTECTED]/16386' At this time, the caller goes to the start (s) extension and keep trying to call somebody up to hangup. I'm currently using the g729a downloaded from Digium's FTP server and Asterisk version (Asterisk CVS-05/30/04-16:28:04 built by [EMAIL PROTECTED] on a i686 running Linux). When I use g711 instead of g729 I have no problem doing that.. Can it be some kind of g729A problem? Did anybody face the same problem? thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI + g729A
Hi Philipp, I'm not receiving my emails sent to the list. I thought my email was not in the asterisk-users list anymore. But anyways.. About the DIAL application, I'm currently use the DIAL application from an AGI program to connect call to Zap channels (i.e, the caller calls an DID number, I do some checks and then call and atendent who is connect to a Zhone channel bank). This setup works fine. The problem is that when I receive calls from IAX2 using g729A, I do the same checks and then call the atendent. But when my AGI program execute the DIAL application, Asterisk says me the it could not create the Zap channel as follows: -- Executing Goto([EMAIL PROTECTED]/16386, 100|1) in new stack -- Goto (default,100,1) -- Executing AGI([EMAIL PROTECTED]/16386, exm) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exm -- AGI Script Executing Application: (DIAL) Options: (Zap/23|12|m) Jun 7 13:28:30 NOTICE[730128]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- AGI Script exm completed, returning 0 -- Hungup '[EMAIL PROTECTED]/16386' Using g711, I have no problem with it.. Do you recommend me to use the old g729b instead the g729a? Better: do you know the difference between them? I were using the g729b, but after a license upgrade, I've started to use the g729a.. regards Oz On Jun 8, 2004, at 10:31 AM, Philipp von Klitzing wrote: Hi there, I don't think you can issue a DIAL statement from within AGI. You'll need to return to the dialplan instead, possibly using some variables that you've set in your AGI script. Also you might want to look at the new application DeadAGI(). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI + g729A
Hello I have the follow situatuion: ISDN | | V E100P || IAX2 / g729A || T100P | Asterisk1 |- - - - - - - - - - - - - - | Asterisk2 | - - - - - - |--| | | | | | Zhone| - - --- Here's the situation: I receive calls from the PSTN in Asterisk1 and forward the call to Asterisk2 (which is connected to a Zhone 100 channel bank). In Asterisk2 I have an AGI application written in C which does some checks before dial one of the extensions. From the AGI program, after done all I have to do, I need to call one extension, but when I call the Dial application, my AGI program exits and Asterisk gives me: -- Executing Goto([EMAIL PROTECTED]/16386, 100|1) in new stack -- Goto (default,100,1) -- Executing AGI([EMAIL PROTECTED]/16386, exm) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exm -- AGI Script Executing Application: (DIAL) Options: (Zap/23|12|m) Jun 7 13:28:30 NOTICE[730128]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- AGI Script exm completed, returning 0 -- Hungup '[EMAIL PROTECTED]/16386' At this time, the caller goes to the start (s) extension and keep trying to call somebody up to hangup. I'm currently using the g729a downloaded from Digium's FTP server and Asterisk version (Asterisk CVS-05/30/04-16:28:04 built by [EMAIL PROTECTED] on a i686 running Linux). When I use g711 instead of g729 I have no problem doing that.. Can it be some kind of g729A problem? Did anybody face the same problem? thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 188 and fax
Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: got (-2) from Queue
Nobody had some expirience with it? Begin forwarded message: From: Osvaldo Mundim [EMAIL PROTECTED] Date: March 9, 2004 6:28:04 PM GMT-03:00 To: [EMAIL PROTECTED] Subject: got (-2) from Queue Hi all, When I call Queue application from AGI, I always got (-2) as returned value. Seeing the show application Queue description, it says that Queue application only returns (-1) and (0). I also tried to see app_queue.c and I cant understand when it happens. Can anybody tell me what this? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] got (-2) from Queue
Hi all, When I call Queue application from AGI, I always got (-2) as returned value. Seeing the show application Queue description, it says that Queue application only returns (-1) and (0). I also tried to see app_queue.c and I cant understand when it happens. Can anybody tell me what this? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] got (-2) from Queue
Hi all, When I call Queue application from AGI, I always got (-2) as returned value. Seeing the show application Queue description, it says that Queue application only returns (-1) and (0). I also tried to see app_queue.c and I cant understand when it happens. Can anybody tell me what this? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] flash button
Hi, Is there a way to control the flash timing in Asterisk? I'm using Siemens euroset 805S analog phones with Asterisk I can transfer a call just hitting a little slower on the on-hook button. The flash button is not working. I was trying to set in zapata.conf changing values of flash and rxflash and it did not work. Is there an other way to do this? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button
That worked perfectly!!! I've adjusted the ZT_MINPULSETIME to 4ms and its working fine! thank you best regards Osvaldo On Mar 2, 2004, at 3:21 PM, Pascal Le Bail wrote: Osvaldo Mundim wrote: Is there a way to control the flash timing in Asterisk? I'm using Siemens euroset 805S analog phones with Asterisk I can transfer a call just hitting a little slower on the on-hook button. The flash button is not working. I had exactly the same problem with exactly the same phone ;-) When I pressed the flash button, Asterisk interpreted the event as a pulse-dialed 1. I looked into zaptel.h and found the following line: #define ZT_MAXPULSETIME (150 * 8)/* 150 ms maximum */ Every pulse shorter than that value is treated as a pulse-dial pulse. Since all my phones seem to generate 100 ms flash pulses, I reduced the 150 to 75 and recompiled Zaptel Asterisk. The problem is solved - and pulse-dialing still works. regards, Pascal Le Bail, Vienna, Austria, Europe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mac X-Lite and Asterisk
Hi Ryan.. No, I don't. I can see just the red one going up and down with my voice. Strange is that on the installation test for mic and speaker, everything was right. Anyways, I will try the SJPhone I see how it would be... Thank you best regards Osvaldo On Feb 19, 2004, at 11:44 AM, Ryan wrote: Do you see the green mic 'audio meter' on X-Lite moving up and down with your voice? I found X-Lite's audio-in to work only intermittently for me with my internal mic, and not at all with my iSight mic. Eventually I opted to use SJ phone, which is working properly. http://www.sjlabs.com/products/sjp-x.html Ryan On 19-Feb-04, at 6:32 AM, Osvaldo Mundim wrote: Mark, My Transmit Silence was already Yes. And an other information is that I'm trying to call a Zap extension on this Asterisk box (using Zhone). From the X-Lite, I can place the call, hear the Zap extension ringing but when I other end answer, we cant hear each other. From the Zap extension, the same thing... Do think is there an other thing which I can try? Osvaldo On Feb 19, 2004, at 10:12 AM, Mark Messmore, Technical Support, University Telcom Inc. wrote: I had this with X-lite (on windows though) where I could hear it on one end but not on the other. On the end where I couldn't hear audio I did this Advanced System Settings -- Audio Settings -- Silence Settings -- Transmit Silence -- Change this to Yes That worked for us. Give that a shot. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Osvaldo Mundim Sent: Thursday, February 19, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mac X-Lite and Asterisk Hi, I'm trying to set up the X-Lite softphone on a Mac Panther and when I try place a call I have no audio on both ends. i.e, I cant hear the other person and same for him. X-Lite is telling me that it has established an connection with Asterisk using GSM codec but stills not transmiting audio. And GSM is allowed on my sip.conf configuration. On this Asterisk box I already have 4 Cisco ATA 188 using g729 working perfectly. Did somebody have the same problem? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mac X-Lite and Asterisk
Hey Ryan, I got the same problem with the SJPhone. Is there something different on the sip configuration for softphone? I'm using: allow=gsm [1234] type=friend insecure=yes nat=yes username=1234 callerid=Osvaldo secret=xx host=dynamic canreinvite=no qualify=200 context=default Osvaldo On Feb 19, 2004, at 11:44 AM, Ryan wrote: Do you see the green mic 'audio meter' on X-Lite moving up and down with your voice? I found X-Lite's audio-in to work only intermittently for me with my internal mic, and not at all with my iSight mic. Eventually I opted to use SJ phone, which is working properly. http://www.sjlabs.com/products/sjp-x.html Ryan On 19-Feb-04, at 6:32 AM, Osvaldo Mundim wrote: Mark, My Transmit Silence was already Yes. And an other information is that I'm trying to call a Zap extension on this Asterisk box (using Zhone). From the X-Lite, I can place the call, hear the Zap extension ringing but when I other end answer, we cant hear each other. From the Zap extension, the same thing... Do think is there an other thing which I can try? Osvaldo On Feb 19, 2004, at 10:12 AM, Mark Messmore, Technical Support, University Telcom Inc. wrote: I had this with X-lite (on windows though) where I could hear it on one end but not on the other. On the end where I couldn't hear audio I did this Advanced System Settings -- Audio Settings -- Silence Settings -- Transmit Silence -- Change this to Yes That worked for us. Give that a shot. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Osvaldo Mundim Sent: Thursday, February 19, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mac X-Lite and Asterisk Hi, I'm trying to set up the X-Lite softphone on a Mac Panther and when I try place a call I have no audio on both ends. i.e, I cant hear the other person and same for him. X-Lite is telling me that it has established an connection with Asterisk using GSM codec but stills not transmiting audio. And GSM is allowed on my sip.conf configuration. On this Asterisk box I already have 4 Cisco ATA 188 using g729 working perfectly. Did somebody have the same problem? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 license
Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
The problem is that I have 2 licenses of 8 channels. One is being used in one of my boxes and the other one is not. What I want is to be sure that the one which I will use in a new Asterisk box is not the one which is being used... Any suggestion? regards Osvaldo On Feb 16, 2004, at 11:57 AM, Wes Marderness wrote: When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA call
Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz 8 headers, 0 lines Retransmitting #1 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=4bd7a841 Content-Length: 0 290Ñ to 200.167.103.219:1025 Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=4bd7a841,uri=sip:[EMAIL PROTECTED] .77,response=1ecb99d4d5e23be179a9eb55eb33c62a Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3642 3642 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:200.167.103.219:1025 SIP/2.0 Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f From: asterisk sip:[EMAIL PROTECTED];tag=as5566fcc8 To: sip:200.167.103.219:1025 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=514a024a,uri=sip:[EMAIL PROTECTED] .77,response=adb7da64c3f557d1db20b699c04f6d84 Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3692 3692 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as046b1041 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=6512ffab Content-Length: 0 to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Retransmitting #1 (no NAT): OPTIONS
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN Just one thing which I did not mention on the last email is that my ATA is behing NAT. Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Some times the sip show peers shows me: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN and some times shows me: Name/usernameHost Mask Port Status porto/porto 200.167.103.219 (D) 255.255.255.255 1025 LAGGED (815 ms) Does the port supposed to be 5060? Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GotoIfTime help
Hey All,
I need to forward an extension to an other depending on the current
time but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten = 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
exten = 111,2,Dial(${Person1})
exten = 111,3,Dial(Hangup)
exten = 333,1,Dial(${Person2})
exten = 333,2,Dial(Hangup)
When I ring on the extension 111, the call is not being forward to the
extension 333..
And the extensions are all in the same context.
regards
Oz
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Re: [Asterisk-Users] GotoIfTime help
All right...
Its working now.
thank you very much!
regards
Oz
On Dec 19, 2003, at 3:02 PM, Philipp von Klitzing wrote:
Hi!
gotoif usually takes a priority as label, not an extension! See below.
Never tried if you can also use the notation
context,extension,priority
instead of just priority, but it might work. Just try it.
I need to forward an extension to an other depending on the current
time but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten = 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333)
exten = 111,2,Dial(${Person1})
exten = 111,3,Dial(Hangup)
exten = 333,1,Dial(${Person2})
exten = 333,2,Dial(Hangup)
exten = 111,1,GotoIfTime(8:00-14:00|*|*|1-12?4:2)
exten = 111,2,Dial(${Person1})
exten = 111,3,Dial(Hangup)
exten = 111,4,Goto(default,333,1)
exten = 333,1,Dial(${Person2})
exten = 333,2,Dial(Hangup)
Note: If you use goto() or gotoif() with just one label then you'll
see a
warning in /var/log/asterisk/messages about the 2nd label missing.
That's
why I prefer to always specify both labels.
Cheers, Philipp
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Re: [Asterisk-Users] E1 in Brazil
Hi, I've asked them about the switch and they told me that its a Siemens EWSD.. Regards Oz On 9/25/03 11:32 AM, Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: Hi. Do you know what switch your telco has ? The one they are using to provide you the service. Osvaldo Mundim Junior wrote: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
I'm sorry, but I don¹t know. I'll ask them and I let you know. Oz On 9/25/03 11:32 AM, Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: Hi. Do you know what switch your telco has ? The one they are using to provide you the service. Osvaldo Mundim Junior wrote: Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as coding. This RDSI are 30 channels completely digital which uses 128k per channel (2Mb). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
Yes, it does. Eduardo was right. They call RDSI as ISDN in Brazil. And its working with an E100P. regards Oz On 9/25/03 11:59 AM, Andrew Kohlsmith [EMAIL PROTECTED] wrote: RDSI and ISDN are the same thing. RDSI is ISDN said in portuguese. The E100P does not do ISDN, does it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call volume on ATA 188
Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not under the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high volume on my side? Tks in advance! Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Call volume on ATA 188
Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not under the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high volume on my side? Tks in advance! Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
